similar to: Re-2: Some questions for chan_capi

Hello list, I have some trouble with chan_capi and asterisk. I'm using 2 Fritz!Card PCI in one * box and testing it with soft phone like sj-phone and x-lite without any serious problems, but: + I don't understand some settings. - what's deflect in capi.conf for? - how to configure multiple isdn-cards in capi.conf? + Some probs with the extensions.conf - exten =>

PLEASE CONFIGURE YOUR AUTORESPONSERS TO NOT SEND MESSAGES TO PEOPLE WHO POST IN MAILING LISTS YOU SUBSCRIBE TO. This is an extremely rude thing to allow, and is becoming increasingly common, especially with users of the Asterisk-Users list. Daryl _____ From: s.speckenheuer@posservice.de [mailto:s.speckenheuer@posservice.de] Sent: Friday, February 04, 2005 6:41 PM To: Daryl G.

Hi C F no asterisk and sip device are not behind same router. actually both are in different countries. how ever when caller and callee are behind same routers voice is just fine (both ways) and i can see re-INVITEs too. but when someone calls from another router then this issue arises. caller can hear the called party but called party can not hear caller. and there are no re-invites issued

supposed to be able to give you much help with such little info anyway), I can only guess that since you are using the 's' extension, you are in a macro ? If so, try scrolling down the wiki page to the example using '[macro-inbound]'.<br> <br> Rob<br> <br> Jonas Kellens wrote: <blockquote cite="mid:4C17C4A1.8020404 at telenet.be"

> Thanks for the reply. > > 1. My VAD is turned off (00140014), and it didn't help for that cut-off. I > am not sure if OutboundProxy has to be configured to have it working fine. > Or this just happened to me? What is your ATA's software? > > 2. I tried dtmfmode=inband on sip.conf, and dtmfmode=rfc2833. None worked. > As per ATA, it is by default using rfc2833.

I have looked at several IAX and SIP soft phones but I have been disappointed with the sound quality on my Windows XP Pro PC. Also the GrandStream problem is that they don't yet support headsets. When I turn auto answer on and I dial in it instantly picks up with the speaker phone. But if I have the handset picked up when a call is coming in the line is busy. That means that the phone itself

Funny - I downloaded the latest Asterisk CVS, and it's pretty much working. Will report when I have some more success. PaulH -----Original Message----- From: Philipp von Klitzing [mailto:klitzing@pool.informatik.rwth-aachen.de] Sent: Tuesday, 28 September 2004 9:46 PM To: Paul Hales Subject: Re: [Asterisk-Users] Leader IP10S Hi! > I have been lent a Leader IP10S phone (SIP) for

Dear all, Our institute has an open position for a Ph.D. student on modeling gene network in stem cells. here is the official announcement: --- The Institute for Biomedical Engineering, Department of Cell Biology, RWTH Aachen University Medical School and Helmholtz Institute for Biomedical Engineering, RWTH Aachen University invites applications of highly motivated individuals for a PhD

Hi! i hope that the right address. my problem is: i'm using a freebsd-server with postfix and sasl > pam to authentificate users via smtp_auth over a windows-pdc from the syslog, i get the message: User "Michael" granted access, also all wbinfo-command work, my e-mail-programm says: 435 error: authentification failed my debug.log tells me: group, endgrent, not found Apr 20

Hi! i hope that the right address. my problem is: i'm using a freebsd-server with postfix and sasl > pam to authentificate users via smtp_auth over a windows-pdc from the syslog, i get the message: User "Michael" granted access, also all wbinfo-command work, my e-mail-programm says: 435 error: authentification failed my debug.log tells me: group, endgrent, not found Apr 20

Does Asterisk support Radius accounting?.... -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] En nombre de asterisk-users-request@lists.digium.com Enviado el: Lunes, 17 de Noviembre de 2003 12:08 p.m. Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users digest, Vol 1 #1912 - 11 msgs Send Asterisk-Users mailing list

Am trying to update my wine i get the following .. [] download]# yum update wine* Loading "priorities" plugin Loading "fastestmirror" plugin Loading mirror speeds from cached hostfile * epel: ftp.nluug.nl * rpmforge: fr2.rpmfind.net * base: ftp.halifax.rwth-aachen.de * updates: ftp.halifax.rwth-aachen.de * addons: ftp.halifax.rwth-aachen.de * extras:

Hi all, I have a cologne chip card which is connected directly to the ntba. Outgoing and incoming calls work fine, but incoming calls from ntba have the wrong callerid (first 0 is missing). I'm using current jolly misdn drivers and chan_misdn-14_04_05 with asterisk stable. Is anyone seeing this behaviour too? Thanks in advance Christian

Hi, I installed Asterisk with CAPI support. Everything works fine while starting Asterisk, but when a call comes in Asterisk hangsup the call after two times of ringing. The output is like: Jun 24 22:19:49 NOTICE[1082178480]: chan_capi.c:1931 capi_handle_msg: CONNECT_IND ID=002 #0x011d LEN=0048 Controller/PLCI/NCCI = 0x101 CIPValue = 0x10

Hello, I have a problem with my Samba4 AD-Controller and NFS. We want to distribute a NFS-Share for our home directory. The problem is that the client is trying to create it on the fly for the first user login but you get 'Permission denied' because of the root_spuash option of the NFS-Server. My question: Is there any posibillity that the normal home directory is created after adding

Hi all ! We have 3 NTBAs which are all going to our existing PBX. Our areacode is 06003 and our DDI enabled number 9141. I want to exchange that PBX with Asterisk, but still struggle to get it working. My CAPI.CONF is currently like this: [general] nationalprefix=0 internationalprefix=00 rxgain=0.800000 txgain=0.800000 [interfaces] context=in-capi msn=60039141 incomingmsn=* controller=1

Hi all, I recently gave LLD a try and it definitely works fine. However one cannot build it together with LLVM_LINK_LLVM_DYLIB: ELF/Driver.cpp and ELF/DriverUtils.cpp pull in llvm/Support/CommandLine.h which defines the command line options so these global variables end up in libLLVM-4.0svn.so via liblldELF. If this shared library is then linked into bin/lld or bin/opt one gets errors because of

I've dealt with similar issues on the LLDB side, I think I can help you with this. pl On 6 January 2017 at 09:39, Rui Ueyama via llvm-dev <llvm-dev at lists.llvm.org> wrote: > Thanks for the info. I can reproduce the issue. But because of lack of cmake > knowledge, I don't know how to fix that now. If no one will take a look at > this, I'll investigate it. > > On

1. You don't appear to be including anything that is within the colon dirs so they are being left out by --prune-empty-dirs. 2. Includes only override excludes that follow them. Since your debug excludes are exceptions to the openSUSE_Leap_42.2/*** include they must be before it not after it. On 01/10/2018 05:46 PM, Istvan Gabor via rsync wrote: > Dear Kevin: > > ~ 1 year ago

Yes of course. We use Samba4 ADDC with winbind to get unix attributes from the DC to the clients. But every time we creating a new member in the AD we have to change the default shell what is /bin/sh to /bin/tcsh. So we want to change the default value of the field in the AD that we don't have to change it every time. The users logging in on a unix domain member. No login on DC